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LINK IP 340P User Manual Please find the latest version of the manual and firmware at : www.linkcom.fr 1/81 ...
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Safety Notices Please read the following safety notices before installing or using this phone. They are crucial for the safe and reliable operation of the device. Please use the external power supply that is included in the package. Other power supplies may cause damage to the device, affect the behavior or induce noise. Before using the external power supply, please be sure it is for use with your power voltage. Incorrect power voltage may cause fire and damage. Please do not damage the power cord. If the power cord or plug is damaged, do not use it. This may cause fire or electric shock. The power plug should be accessible at all times because this is the only way to remove power from the device. Handle the phone carefully. Do not drop it or shake it. Rough handling can cause internal damage. Do not install the device in direct sunlight. Also do not put the device on carpets or cushions, or other poorly ventilated locations. This may cause fire or overheating. Avoid exposure to temperatures above 40 , below 0 or high humidity. Avoid wetting the unit with any liquid. Do not use harsh chemicals, cleaning solvents, or strong detergents to clean the device. ...
Introducing Link IP 340P Thank you Thank you for purchasing the Link IP 340P Over Internet Protocol (VoIP) telephone. The IP 340P is a fully featured telephone that provides voice communication over the data network. This phone has all the features of a traditional telephone and all gives access to many data service features. This guide will help you easily use the various features and services available on your phone. Box Contents The following items should be packed with your telephone. Please contact your dealer if any of them are missing. Telephone (Main body) with display and keypad Handset Handset cord Power supply Ethernet cable Keypad Key Key name Function Description These keys are used in many areas of phone operation. Navigation Depending on the application they will have different functions.
Volume ‐/+ Adjust the volume by pressing these two keys. Speaker Activate speakerphone mode. Indicator This light blinks to indicate a missed call. light Various functions depending on the phone mode. Soft Description will be shown in LCD. key 1/2/3/4 Keyboard Dial phone numbers Input/Output Ports Port Port name Description Power switch Input: 5V AC, 1A WAN 10/100M Connect it to Network LAN 10/100M Connect it to PC Handset Port type: RJ‐9 connector Icon Introduction ...
Call hold Auto answer Contact DND(Do not Disturb) In hand free mode In handset mode SMS Missed call Call forward LED Introduction 1.6.1 Power Indication LED (Power Light Enabled) LED Status Description Steady red Power on. Blinking red There is an incoming call. Off Power off. 1.6.2 Power Indication LED (Power Light Disabled) LED Status Description Blinking red There is an incoming call Initial Connection and Setting Connecting the phone Connect to the network. Use the Ethernet cable in the package to connect the WAN port on the back of your phone to an Ethernet port. The following 3 figures show connection options. ...
Direct network connection—Use this method if you have a single Ethernet port which is already in use. Disconnect the Ethernet cable from the Ethernet port and attach it to the WAN port on the back of the phone. Then use the Ethernet cable in the package to connect the LAN port on the back of the phone to the other device. The IP Phone now shares a network connection. Access by router connection—Connect one end of the network cable to the IP 340P’s WAN port the other end is connected to your broadband router’s LAN port, so that the completion of the network hardware connections. In most cases, you must configure your network settings to DHCP mode. 2. Connect the handset to the handset jack using the handset cable in the package. 3. Connect the power supply to the DC port on the back of the phone. Connect the power supply to a standard power outlet. Note that the power supply will not be needed if your network provides Power over Ethernet (PoE). 4. The phone’s LCD screen displays “INITIALIZING”. Later, a ready screen displays the date, time and current network mode. If your LCD screen displays different information from the above, more information may need to be entered. Please refer to the next section. If your phone registers into your IP telephony Server, it is ready to use. If not, continue to read for more configuration information. Network Settings DHCP is supported by default. This allows the phone to receive an IP address and other network‐related settings (Netmask, IP gateway, DNS server) from the DHCP server. If no DHCP server is available, the network connection settings must be changed. Follow the instructions below to change to either PPPoE or static IP. 10/81...
2.2.1 PPPoE Mode 1. Press the Menu softkey. 2. Scroll down to “3 Settings.” 3. Press Enter. 4. Scroll down to “2 Advanced Settings.” 5. Press Enter. 6. The LCD will display “Enter Password”. 7. Input the password (default value is 123). 8. Press Eenter. 9. Scroll down to “2 Network.” 10. Press Enter. 11. Press Enter to select WAN Settings. 12. Scroll down to “4 PPPoE Settings.” 13.
10. Press Enter to select WAN Settings. 11. Scroll down to “2 Static IP Settings.” 12. Press Enter. 13. Use the keypad to enter the IP Address. 14. Press Save softkey. 15. Press Down key. 16. Use the keypad to enter the Subnet Mask. 17. Press Save softkey. 18. Press Down key. 19. Use the keypad to enter the Gateway Address. 20. Press Save softkey. 21. Press Down key. 22. Use the keypad to enter the DNS 1 Address. 23.
18. Press Save softkey. 19. Press Back softkey. 20. Press up/down key to scroll to “1 Connection Mode.” 21. Press Enter. 22. Use vol‐/vol+ to select “DHCP.” 23. Press Save softkey. 24. Press Back or Exit 6 times to return to idle screen. 25. Disconnect and reconnect the power supply so the phone will reboot and apply the new settings. Basic Functions Making a call 3.1.1 Call Device Calls can be made using two different devices: 1. Handset ‐ Pick up the handset. The icon will be shown on the LCD screen. 34. Speakerphone ‐ Press the Speaker button. The icon will be shown on the LCD screen. ...
If you select Disabled,Incoming calls will be ring and stored in the Call History. Call Forward This feature allows forwarding an incoming call to another phone number. The display shows icon. The following call forwarding events can be configured: Off: Call forwarding is deactivated by default. Always: Incoming calls are immediately forwarded. Busy: Incoming calls are immediately forwarded when the phone is busy. No Answer: Incoming calls are forwarded when the phone is not answered after a specific period. To configure Call Forward via Phone interface: 1. Press Menu ‐>Features‐>Enter>Call Forward‐>Enter. 2. Select the line to be forwarded. 3. Use vol‐/vol+ to select Disabled, Always, Busy, or No Answer. 4. After choosing a mode (except Disabled), press Down key and then enter the phone number for forwarding. 5.
transferred followed by "#" and press Send. After the third party answers, press XFER to complete the transfer. NOTE: Call waiting and call transfer must be enabled. NOTE: The SIP server must support RFC3515. 3.7.3 Semi‐Attended Transfer During a conversation, press the XFER key, dial the number to which the call is to be transferred. Then press the Send softkey. When the third party phone begins to ring, press XFER to complete the transfer. NOTE: Call waiting and call transfer must be enabled. 3‐way conference call 1. Press the Conf softkey during an active call. 39. The first call will be placed on hold and dial tone will be heard. 40. Dial the number to be added to the conference. 41. Press Dial. 42. When the call is answered, press Conf to add the caller to the conference. 43. To release the conference, press Split. Multiple‐way call To add a fifth party to four active calls 1. Press Conf softkey or XFER softkey 2. Press OK 3. Enter the number 4. Press Dial and wait for the other party to answer. 5.
Join call This allows a third party to join an existing call. For example: If B and C are on a call, A can join by dialing a code plus the number for B or C. This assumes that B or C also support Join Call. The following chart shows how to configure this in the dial peer screen. *2* is the code. After saving the above configuration, A can dial *2* plus the number for B or C to join B and C’s call. The prefix can be set to anything the user desires that does not interfere with other dialing rules. Redial / Unredial If B is on a call when A calls, A will get busy tone. If A wants to connect to B as soon as B is available, he can use the redial function. To use this feature, A dials a prefix and then B’s number. When the redial function is activated, A will check B’s calling status every 60 seconds. When B is available, A’s phone will ring. When A goes off hook, the phone will call B automatically. If A does not want to call B, the redial function can be cancelled by dialing a prefix plus B’s number. *3* is the redial prefix code. A can dial *3* plus B’s phone number to activate the redial function. *4* is the unredial prefix code. A can dial *4* to cancel the redial function. The user can select any prefix as long as it does not interfere with dialing rules. ...
Hotline/Warmline This feature will cause the phone to place a call to a programmed number whenever it goes off‐hook. A different hotline number can be set for each SIP line. Speed dial This feature will allow you make speed dial easily. If you set up speed dial with name and tel numbers for 1~9, and then you can dial n# to make the corresponding speed dial number directly. Application 4.9.1 SMS 1. Press + ‐>Applications‐>Enter‐>SMS‐>Enter. 2. Use the navigation keys to highlight the options. Messages can be read in the Inbox/Outbox. 3. Press Reply to reply to a message. Use the 2aB softkey to change the Input Method. After entering the reply, press OK, use the navigation keys to select the line from which you want to send, then press Send. 4. To write a new message, press New. Use the 2aB softkey to change the Input Method. After entering the reply, press OK, use the navigation keys to select the line from which you want to send, and press Send. 5. To delete a message, press Del. You have three options to choose: Yes, All, No. 4.9.2 Memo Memos can be recorded in the phone as reminders. Press Menu‐>Application‐>Memo‐>Enter‐>Add. Options for Mode, Date, Time, and Ring Tone can then be configured. The reminder text can also be entered. ...
4.9.4 Ping 1. Press Menu‐>Application‐>Ping‐>Enter. 2. Enter the IP Address to be pinged. 3. Press Start 4. Display will show “Ping IP Address” 5. After approximately 5 seconds, the display will show “OK” if the ping is successful or “Failed” is the ping is unsuccessful. Other Functions Call Forward If this feature is enabled, the phone will forward to another phone. 6. Press Menu ‐>Features‐> Enter‐>Auto Answer‐> Enter. 7. Use Up/Down key to select line. 8. Use vol‐/vol+ to Enable. 9. Use Up/Down key to access number setting. Auto Answer If this feature is enabled, the phone will answer a ringing line after a specified time. Press Menu ‐>Features‐> Enter‐>Auto Answer‐> Enter. Use Up/Down key to select line. Use vol‐/vol+ to Enable. Use Up/Down key to access time setting. Use keypad to enter time in seconds. Auto Handdown This is the time after a call ends before the phone returns to the idle state. 1. Press Menu ‐>Features‐> Enter‐>Auto Handdown‐> Enter. 2. Use vol‐/vol+ to Enable. 3. Use Up/Down key to access time setting. 4.
DND If this function is enabled the new incoming calls will be rejected. Press Menu ‐>Features‐> Enter‐>DND‐> Enter. Use vol‐/vol+ to Enable. Use Up/Down key to access line setting. Use vol‐/vol+ to Enable. Ban Anonymous If this function is enabled, the phone will block calls with no Caller ID information. 1. Press Menu ‐>Features‐> Enter‐>Ban Anonymous Call‐> Enter. 2. Choose the SIP Account from which to Ban Anonymous Call. 3. Press OK 4. Use vol‐/vol+ to Enable. Ban Outgoing If this function is enabled, the phone cannot make outgoing calls. Press Menu ‐>Features‐> Ban Outgoing‐> Enter. Hotline If you turn on automatically as you set the number of call setup time. Press Menu ‐>Features‐> Enter‐>Hotline‐> Enter. Use Up/Down key to select line. Use vol‐/vol+ to Enable. Use Up/Down key to access time setting. Use Up/Down key to access number setting. Dial Plan 1. Press Menu ‐>Features‐> Enter‐>Dial Plan‐> Enter. 2. The following items in the dial plan can be enabled or disabled: Press # to Send, Timeout to Send, Timeout, Fixed Length Number, Press # to Do BXFER, BXFER On Onhook, AXFER On Onhook. Note: It is recommended that Dial Plan be configured from the web interface. 5.10 Dial Peer 1.
5.11 Intercom Enables/Disables Intercom calls Press Menu ‐>Features‐> Enter‐>Intercom‐> Enter. 5.12 Auto Redial If Auto Redial is enabled, the phone will continue to retry a busy call. The user sets the retry interval and the number of times to redial. The user is also given the option to activate this feature on each busy call. 1. Press Menu ‐>Features‐> Enter‐>Auto Redial‐> Enter. 2. Use vol‐/vol+ to Enable. 3. Use Up/Down key to select Interval and Times. 4. Press Save. 5.13 Call completion This is similar to Auto Redial except that it detects the state of the called number before making a new call attempt. 1. Press Menu ‐>Features‐> Enter‐>Call Completion‐> Enter. 2. Use vol‐/vol+ to Enable. 3. Press Save. 5.14 Power Light his feature enables the power light at the bottom of the phone. Press Menu ‐>Features‐> Enter‐>Power LED‐> Enter. 5.15 Hide DTMF This feature sets how DTMF digits are displayed after a call is in progress. For example, dial a PIN code to access banking information. 1.
be set. Example: A call is placed to 6625551212. Password is set to 662 and length is set to 3. Display will show 662***1212. 1. Press Menu ‐>Features‐> Enter‐>Passwd Dial‐> Enter. 2. Use vol‐/vol+ to enable the feature. 3. Use Up/Down key to move to Prefix. 4. Use keypad to enter prefix. 5. Use Up/Down key to move to Length. 6. Use keypad to enter Length. 7. Use BACK or EXIT to return to idle screen. 5.17 Pre Dial If this feature is enabled, digits dialed on‐hook will be transmitted when the phone goes off‐hook Press Menu ‐>Features‐> Pre Dial‐> Enter. 5.18 Call Logs If this feature is disabled,you will not see the call logs. Press Menu ‐>Features‐> Enter‐>Call Logs‐> Enter. Use vol‐/vol+ to enable. 5.19 Default Line If this feature is disabled The handset displays Greeting Words. Press Menu ‐>Features‐> Enter‐>Call Logs‐> Enter. Use vol‐/vol+ to enable. 5.20 Auto Switch Line If this feature is enabled,then the opportunity to use the first available line call path. ...
held. Soft Key – Keys under the display 3. Use Up/Down key and Enter to select the key. 4. Use vol‐/vol+to select the function. 5. Press OK to save. 6. Use BACK or EXIT to return to idle screen. Screen Settings 1. Press Menu ‐>Settings‐> Enter‐>Basic Settings‐> Enter‐>Screen Settings‐>Enter. 2. The following items can be set. Contrast – Set the contrast of the LCD. Contrast Calibration – Set the level of contrast that the current contrast setting provides. Backlight – Enable or disable LCD backlight. 3. Press OK to save. 4. Use BACK or EXIT to return to idle screen. Ring Settings 6.3.1 Ring Volume 1. Press Menu ‐>Settings‐> Enter‐>Basic Settings‐> Enter‐>Ring Settings‐>Enter‐>Ring Volume‐>Enter. 2. Use vol‐/vol+ to select the desired ring volume from the 9 choices. The phone will ring at the selected volume shortly after it is selected. 3. Press Save. 4. Use BACK or EXIT to return to idle screen. 6.3.2 Ring Type ...
Time & Date 1. Press Menu ‐>Settings‐>Enter‐>Basic Settings‐> Enter‐>Time & Date‐>Enter. 2. Use vol‐/vol+ to choose Auto or Manual. If Auto is chosen, the phone will get date and time information from a time server. The IP address of this server may need to be entered. If Manual is chosen, the date and time must be entered. 3. Use Up/Down key to move to the following items. Use vol‐/vol+ to make selection. SNTP Server – Time Server IP address – This is the only item that must be configured if auto is chosen. Time Zone – This is shown as an offset from GMT. Format – Date Display format. Type – Character used as delimiter in date display. 12 Hour Clock – If disabled, clock is 24 hour. Daylight Saving Time 4. Press Save. 5. Use BACK or EXIT to return to idle screen. Greeting Words This feature shows the words displayed in the upper left of the LCD. Default is VOIP PHONE. 1. Press Menu ‐>Settings‐> Enter‐>Basic Settings‐> Enter‐>Greeting Word‐>Enter. 2. Enter the message using the keypad. It may be necessary to change the input mode using the soft keys. Use DELETE to remove characters and 0 for space. Maximum message length is 12 characters. ...
2. Outbound Proxy ‐ SIP Outbound Proxy IP Address 3. Registration – Enable or disable registration for this account. 4. Server Address – SIP Server IP Address 5. Server Port – SIP Port – Default 5060 6. SIP User – SIP User name 7. Auth User – User name for authentication 8. Auth Password – Password for authentication 7.1.2 Advanced Settings 1. Domain Realm – SIP Domain 2. Dial Without Registered – Enable or disable dialing with no SIP registration 3. Anonymous – Privacy Support. Choose RFC3323, RFC3325 or None 4. DTMF Mode – Choose RFC2833, SIP_Info, In‐band, or Auto 5. Use STUN – Enable or disable use of STUN Server. If enabled, the IP address of the STUN server must be entered. 6. Local Port – Local SIP Port – Default 5060 7. Ring Type – Select ring type for this account. See Section 6.3.2. 8. MWI Number – Number for Message Waiting 9. Pickup Number – Code for call pickup 10. Park Number – Code for call park 11.
Maintenance See Section 8.3.6 for a detailed explanation of each option. It is recommended that these features be accessed through the web interface. 4. Auto Provision – Select DHCP Option, Plug and Play, or Phone Flash for autoprovision. 5. TR069 – Enable or disable configuration via TR069. 6. Backup – Select Config, Phonebook or none for backup. File name must be entered. 7. Upgrade – Select Image, MMI Set, BMF, Ring, Config, or Phonebook for upgrade. File name must be entered. Factory Reset Choose Yes to return the phone to factory default settings. Web Configuration Introduction of configuration 8.1.1 Configuration Methods There are three methods which can be used to configure this phone: Phone keypad – As discussed in previous sections Web browser ‐ Recommended way Telnet with CLI command 8.1.2 Password Configuration There are two levels of access: root level and general level. A user with root level access can browse and set all configuration parameters, while a user with general level can set all configuration parameters except server parameters for SIP or IAX2. ...
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Safari are supported browsers. If the IP address is not known, it can be displayed on the phone’s LCD by pressing the Menu‐>Status. After entering the IP address, the following screen is displayed. After configuring the IP phone, remember to click SAVE under the Maintenance tab. If this is not done, the phone will lose the modifications when it is rebooted. 26/81...
Configuration via WEB 8.3.1 BASIC 8.3.1.1 Status Field Name Explanation Network Shows the configuration information for WAN and LAN port, including connection mode of WAN port (Static, DHCP, PPPoE), MAC address, IP address of WAN port and LAN port, DHCP server status for LAN port (ENABLED or DISABLED). Accounts Shows the phone numbers and registration status for the 2 SIP LINES and 1 IAX2 server. 27/81...
8.3.1.2 Wizard Select the appropriate network mode. The phone supports three network modes: Static: The parameters of a Static IP connection must be provided by your ISP. DHCP: In this mode, network parameter information will be obtained automatically from a DHCP server. PPPoE: In this mode, you must enter your ADSL account and password. Refer to Section 2.2 for detailed information about configuring the network parameters. 28/81...
8.3.1.2.1 Static IP If Static IP is selected, this screen will be displayed. Information provided by the ISP should be entered. Click Back to return to the Wizard screen. Click Next to go to Quick SIP Settings 8.3.1.2.2 DHCP After selecting DHCP and clicking NEXT, the Quick SIP Settings screen will appear. Click Back to return to the Wizard screen. Click Next to go to the Summary screen. 8.3.1.2.3 PPPoE If PPPoE is selected, this screen will appear. Enter the information provided by the ISP. Click Back to return to the Wizard screen. Click Next to go to Quick SIP Setting. 29/81...
8.3.1.2.4 Quick SIP Settings Field Name Explanation Display Name The name shown in caller ID. Server Address SIP server address either IP address or URI. Server Port SIP server port (usually 5060). Authentication User Login name or Authentication ID. Authentication Password SIP password. SIP User Phone number. Enable Registration Submits registration information. Normally checked. Click Back to return to the IP Address screen. Click Next to see summary screen. Click Finish button to save settings and reboot. After the reboot, SIP calls can be made. 30/81...
8.3.1.3 Call Log Outgoing call logs can be seen on this page. Field Name Explanation Start Time Start time of the outgoing call Duration Duration of the outgoing call. Dialed Calls Account, protocol, and line of the outgoing call. 8.3.1.4 Language Field name Explanation Language Set the language of phone. English is default. Greeting Words The greeting displayed on LCD when phone is idle. It has a maximum of 12 English characters. Default is VOIP PHONE. 31/81...
8.3.2 Network 8.3.2.1 WAN Config Field Name Explanation Active IP Address The current IP address of the phone. Current Subnet Mask The current Subnet Mask. Current IP Gateway The current Gateway IP address. MAC Address The MAC address of the phone. MAC Timestamp Time the MAC address was obtained. WAN Settings The phone supports three network modes. These are also discussed in Section 2.2. Static: Network parameters must be entered manually and will not change. All parameters are provided by the ISP. DHCP: Network parameters are provided automatically by a DHCP server. PPPoE: Account and Password must be input manually. These are provided by your ISP. 32/81...
8.3.2.1.1 Static IP If Static IP is chosen, the screen below will appear. Enter values provided by the ISP. 8.3.2.1.2 DHCP If DHCP is chosen, all configuration information will be provided by a DHCP server. Contact the ISP to determine if DHCP is used. 8.3.2.1.3 PPPoE If PPPoE is chosen, the screen below will appear. Enter the information provided by the ISP. Service Name IP Address or name of DSL Server User DSL User Name or Login ID Password DSL Password After entering the new settings, click the APPLY button. The phone will save the new settings and apply them. If a new IP address was entered for the phone, it must be used to login to the phone after clicking the APPLY button. 33/81...
8.3.2.2 LAN Config Field Name Explanation IP Address LAN static IP. Subnet Mask LAN Subnet Mask. DHCP Service Activate DHCP server for LAN port. The phone must be rebooted for the DHCP server setting to take effect. NAT Enable NAT operation Port Mirror Port Mirror can only be activated in bridge mode. If activated, the data stream from the WAN port is copied to the LAN port of the phone. Enable Bridge Mode If Bridge Mode is activated, the phone will not provide an IP address for the LAN port. Instead, the LAN and WAN will be part of the same network. If this is activated, clicking Apply, will cause the phone will reboot. Note: When LAN IP or bridge mode status is changed, the system will reboot! If bridge mode is chosen, static LAN configuration will be disabled automatically. 8.3.2.3 Qos & VLAN Config ...
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Chart 1 shows a network switch with no VLAN. Any broadcast frames will be transmitted to all other ports. For example, and frames broadcast from Port 1 will be sent to Ports 2, 3, and 4. Chart 2 shows an example with two VLANs indicated by red and blue. In this example, frames broadcast from Port 1 will only go to Port 2 since Ports 3 and 4 are in a different VLAN. VLANs can be used to divide a network by restricting the transmission of broadcast frames. . Note: In practice, VLANs are distinguished by the use of VLAN IDs 35/81...
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Field Name Explanation Enable LLDP Enable or Disable Link Layer Discovery Protocol (LLDP) Packet Interval The time interval for sending LLDP Packets Enable Learning Function Enables the telephone to synchronize its VLAN data with the Network Switch. The telephone will automatically synchronize DSCP, 802.1p, and VLAN ID values even if these values differ from those provided by the LLDP server. Enable DSCP Enable or Disable Differentiated Services Code Point (DSCP) SIP DSCP Specify the value of the SIP DSCP in decimal Audio DSCP Specify the value of the Audio DSCP in decimal Enable WAN Port VLAN Enable or Disable WAN Port VLAN WAN Port VLAN ID Specify the value of the WAN Port VLAN ID. Range is 0‐4095 SIP 802.1P Priority Specify the value of the voice 802.1p priority. Range is 0‐7 Audio 802.1P Priority Specify the value of the signal 8021.p priority. Range is 0‐7 LAN Port VLAN Mode Follow WAN: LAN Port ID is same as WAN ID Disable: Disable Port VALN Enable: Specify a VLAN ID for the LAN port which is different ...
8.3.2.4 Service Port Set the port values for Telnet/HTTP/RTP on this page. Field Name Explanation Web Server Type Specify Web Server Type – HTTP or HTTPS HTTP Port Port for web browser access. Default value is 80. To enhance security, change this from the default. Setting this port to 0 will disable HTTP access. Example: The IP address is 192.168.1.70 and the port value is 8090, the accessing address is http://192.168.1.70:8090. HTTPS Port Port for HTTPS access. Before using https, an https authentication certification must be downloaded into the phone. Default value is 443. To enhance security, change this from the default. Telnet Port Port for Telnet access. The default is 23. RTP Port Range Start Set the beginning value for RTP Ports. Ports are dynamically allocated. RTP Port Quantity Set the maximum quantity of RTP Ports. The default is 200. Notes: 8. Any changes made on this page require a reboot to become active. 9.
8.3.2.5 DHCP SERVICE Field Name Explanation DHCP Client Table IP‐MAC mapping table. If the LAN port of the phone connects to a device, this table will show its IP and MAC address. Leased Table Name Name of the lease table. Start IP Address Beginning IP address of the lease table. End IP Address Ending IP address of the lease table. A device connected to the LAN port will get an IP address between Start IP and End IP. Subnet Mask Subnet Mask of the lease table. IP Gateway Network Gateway of the lease table. Leased Time Time IP address assignments will persist. Unit is minutes. DNS Server Address IP address of DNS server. Add Click this button to add this lease table DHCP Lease Table Enter the table name and click the Delete button to remove a DHCP Delete lease table. Enable DNS Relay Activates DNS Relay in the phone. Default is enabled. Notes: 38/81...
11. The size of lease table cannot be larger than the quantity of C network IP address. It is recommended to use the default lease table without modification 12. If the DHCP lease table is modified, the phone must be rebooted. 8.3.2.6 TIME&DATE Set the time zone and SNTP (Simple Network Time Protocol) server on this page. Daylight savings time configuration and manual time and date entry are also done on this page Field Name Explanation Simple Network Time Protocol (SNTP) Settings Enable SNTP Enable or Disable SNTP Enable DHCP Time If this is enabled, phone will synchronize time with DHCP server. Primary Server IP address of Primary SNTP Server Secondary Server IP address of Secondary SNTP Server 39/81...
Time Zone Local Time Zone Resync Period Time between resync to SNTP server. Default is 60 seconds. 12 ‐Hour Clock If checked, clock is 12 hour mode. If unchecked, 24 hour mode. Default is 24 hour mode. Date Format Specify the date format. Fourteen different formats are available. Date Separator Four date separators are available: /, ‐ , . , space Daylight Saving Time Settings Enable Enable daylight saving time. Offset(minutes) DST offset. Default is 60 minutes. Month Start and end month for DST Week Start and end week for DST Day Start and end day for DST Hour Start and end hour for DST Minute Start and end minute for DST Manual Time Settings Enter the values for the current year, month, day, hour and minute. All values are required. Note: Be sure to disable SNTP service before entering manual time and date. 8.3.3 VOIP 8.3.3.1 SIP Configuration Configure a SIP server on this page. ...
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Field Name Explanation Choose the sip line to configured (SIP 1 – SIP2). Click the dropdown arrow to select the line. Status Shows registration status. Will show “Registered” if registered or “Unapplied” if not registered. Server Address SIP server IP address or URI. Server Port SIP server port. Default is 5060. Authentication User SIP account name (Login ID). Authentication Password SIP registration password. SIP User Phone number assigned by VoIP service provider. Phone will not register if there is no phone number configured. Display Name Set the display name. This name is shown on Caller ID. Enable Registration Check to submit registration information. Domain Realm SIP Domain if different than the SIP Registrar Server. Proxy Server Address SIP proxy server IP address or URI(This is normally the same as the SIP Registrar Server) Proxy Server Port SIP Proxy server port. Normally 5060. Proxy User SIP Proxy server account. Proxy Password SIP Proxy server password. Backup Server Address Backup SIP Server Address or URI (This server will be used if the ...
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No Ans. Fwd Wait Time Used in conjunction with Call Forward No Answer. Wait time in seconds before call is forwarded. Transfer Timeout Time interval between sending “bye” message and hanging up after the phone transfers a call. Enable Hotline Activate Hot Line feature. Automatically call a number by going off hook. Hotline Number Number to be called in Hot Line Mode. Warm Line Wait Time Used in Hot Line Mode. Time the phone waits after off hook before dialing the hot line number. SIP Encryption Enable/Disable SIP Encryption. SIP Encryption Key SIP Encryption key. RTP Encryption Enable/Disable RTP Encryption. RTP Encryption Key RTP encryption key Enable Auto Answer Activate Auto Answer mode. If activated, phone will automatically answer an incoming call. Auto Answer Timeout Used in conjunction with Auto Answer. The phone will answer an incoming call after the Auto Answer Timeout Enable Session Timer If enabled, this will refresh the SIP session timer per RFC4028. ...
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answer within a designated time. The incoming call record will not be displayed in the Call History. No Ans. CFwd Off Code Disable Server No Ans. CFwd as described above. Ban Anonymous On Code Ban Anonymous On – When this function is enabled, the server will disallow the phone to make anonymous calls. Ban Anonymous Off Code Allow Anonymous Calling function described above. In other words “Anonymous” will be transmitted for Caller ID. Keep Alive Type Specifies the NAT keep alive type. If SIP Option is selected, the phone will send SIP Option sip messages to the server every NAT Keep Alive Period. The server will then respond with 200 OK. If UDP is selected, the phone will send a UDP message to the server every NAT Keep Alive Period. Keep Alive Interval Set the NAT Keep Alive Interval. Default is 60 seconds User Agent Set SIP User Agent value. DTMF SIP INFO Mode You can chose Send 10/11 or Send */# DTMF Type DTMF sending mode. There are four modes: In‐band RFC2833 SIP_INFO AUTO Different VoIP Service providers may require different modes. Local port SIP port. Default is 5060. Ring type ...
for SIP messages above 1500 bytes Enable Strict Proxy Enables the use of strict routing. When the phone receives packets from the server,it will use the source IP address, not the address in via field. Enable GRUU Support for Globally Routable User‐Agent URI (GRUU) Enable Displayname Puts quotation marks around the display‐name in SIP messages. Quote For servers that require this. Enable user=phone Sets user=phone in SIP messages. For compatibility with servers that require this. Click to Talk Set click to Talk (needs support from server). Respond 182 when Call Enable phone responds 182 instead of 180 in some SIP server waiting envrionment Use VPN Enable SIP use VPN for every line individually, not all of them Enable DND Enable DND for SIP line individually SIP Global Settings Strict Branch Enable Strict Branch ‐ The value of the branch must be after “z9hG4bK” in the VIA field of the INVITE message received, or the phone will not respond to the INVITE. Note: This will affect all lines Enable Group Enable SIP Group Backup. This will affect all lines Registration Failure Retry Registration failure retry time – If registration fails, the phone will ...
Field Name Explanation Status Shows registration status. Will show “Registered” if registered or “Unapplied” if not registered. Server Address IAX2 server address. Server Port IAX2 server port. Default is 4569. Account IAX2 account name for registration Password IAX2 registration password. Phone Number IAX2 phone number (usually the same as IAX2 account name). Local Port IAX2 local port. Default is 4569. Voice Mail Number Voice mail number. Voice Mail Text Voice mail name. Echo Test Number If the IAX2 server supports echo test and the echo test number is non‐ numeric, this number can be used to replace the echo test text. This allows dialing a number to perform an echo voice test. This function is provided to test whether communication through the server. Echo Test Text Echo test text Refresh Time Expiration time of IAX2 server registration. Allowed values are between 60 and 3600 seconds. Enable Registration Enable/Disable IAX2 registration. ...
Field Name Explanation STUN NAT Transversal Shows whether or not STUN NAT Transversal was successful. Server Address STUN Server IP address Server Port STUN Server Port – Default is 3478. Binding Period STUN blinding period – STUN packets are sent at this interval to keep the NAT mapping active. SIP Waiting Time Waiting time for SIP. This will vary depending on the network. Local SIP Port Local SIP Port‐ Default is 5060 SIP Line Using STUN SIP Line Using STUN Select the Line for use with STUN (SIP 1 ‐ SIP 6) Use STUN Enable/Disable STUN on the selected line. 8.3.3.4 DIAL PEER This feature allows the user to create rules to make dialing easier. There are several different options for dial rules. The examples below will show how this can be used. Example 1: Substitution – Assume that it is desired to place a direct IP call to IP address 192.168.119. Using this feature, 156 can be substituted for 192.168.1.119. ...
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Example 3: Addition – Two examples are shown. In the first case, it is assumed that 0 must be dialed before any 11 digit number beginning with 13. In the second case, it is assumed that 0 must be dialed before any 11 digit number beginning with 135, 136, 137, 138, or 139. Two different special characters are used. x – Matches any single digit that is dialed. [] – Specifies a range of numbers to be matched. It may be a range, a list of ranges separated by commas, or a list of digits. Field Name Explanation Phone number There are two types of matching: Full Matching or Prefix Matching. In Full matching, the entire phone number is entered and then mapped per the Dial Peer rules. In prefix matching, only part of the number is entered followed by T. The mapping with then take place whenever these digits are dialed. Prefix mode supports a maximum of 30 digits. Destination Set Destination address. This is optional. For a peer to peer call, 48/81...
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enter the destination IP address or domain name. To use a dial rule on the SIP2 line, enter 0.0.0.2. Port Set the Signaling port, the default is 5060. Alias Set the Alias. This is the text to be added, replaced, or deleted. It is optional. Note: There are four types of aliases. 1) Add: xxx – xxx will be dialed before any phone number. 2) All: xxx – xxx will replace the phone number. 3) Del: The characters will be deleted from the phone number. 4) Rep: xxx – xxx will be substituted for the specified characters. Call Mode Select either SIP or IAX2 protocol. Suffix Characters to be added at the end of the phone number. This is optional. Delete Length Sets the number of characters to be deleted. For example, if this is set to 3, the phone will delete the first 3 digits of the phone number. This is optional. Dial Peer Examples Web Interface Explanation Example Set phone number, Dial “93333” Destination, Alias and Delete The SIP2 server will Length. receive “3333” Phone number is XXXT; ...
Set Phone Number, Alias and Dial “0106228” Delete Length. Phone number The SIP1 server will is XXXT and Alias is rep: xxx receive “86106228” If the dialed phone number starts with the digits in the Phone Number box, the matching digits will be replaced by the alias number. If the dialed phone number Dial “147” starts with the digits in the The SIP1 server will Phone Number box, the phone receive “1470011” will send out the dialed phone number and add the suffix number. 8.3.4 Phone 8.3.4.1 AUDIO This page configures audio parameters such as voice codec, handset volume, and ringer . volume Field Name Explanation First Codec The first codec choice: G.711A,G.711A u, G.722, G.723.1, G.729AB, G.726‐32 ...
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Third Codec The third codec choice: G.711A, G.711A u, G.722, G.723.1, G.729AB, G.726‐32,None Fourth Codec The forth codec choice: G.711A,G.711A u, G.722, G.723.1, G.729AB, G.726‐32, None Fifth Codec The fifth codec choice G.711A,G.711A u, G.722, G.723.1, G.729AB, G.726‐32, None Sixth codec The sixth codec choice G.711A,G.711A u, G.722, G.723.1, G.729AB, G.726‐32,None Onhook Time Time the handset must be on hook to disconnect a call. Default is 200ms. Default Ring Type Ring Sound – There are 9 standard types and 3 User types Handset Volume Handset Microphone volume – 9 levels Speakerphone Volume Speaker volume in hands free mode ‐ 9 levels Speakerphone Ring Speaker Ring Volume ‐ 9 levels Volume G729AB Payload G729AB Payload Length – Adjusts from 10 – 60 mSec Length Tone Standard Select tone plan for the country of operation G722 Timestamps Choices are 160/20ms or 320/20ms G723.1 Bit Rate Choices are 5.3kb/s or 6.3kb/s Enable VAD Enable or disable Voice Activity Detection (VAD). If VAD is enabled, G729 Payload length cannot be set greater than 20 mSec. ...
8.3.4.2 FEATURE This page configures various features such as Hotline, Call Transfer, Call Waiting, etc. Field Name Explanation DND might be disabled, phone for all SIP lines, or line for SIP DND(Do Not Disturb) individually. Enable Call Transfer If enabled, Call Transfer is allowed. Semi‐Attended If enabled, Semi‐Attended Transfer is allowed. Transfer Enable Auto Handdown If enabled in speakerphone mode, the phone will automatically hang up and return to idle when the distant party terminates the call. In handset mode, it will play dial tone instead of returning to idle. Auto Handdown Time Wait time before the phone performs the Auto Handdown behavior described above. Enable Auto Redial If enabled, the phone will automatically redial a call if a busy tone is received. Auto Redial Interval Wait time between auto redial attempts in seconds. Auto Redial Times Maximum number of auto redial attempts. Enable Intercom If enabled, allows intercom calls. Enable Intercom Tone ...
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Turn Off Power Light Disables Power Light if selected. Emergency Call The phone will dial the emergency call number even if the keyboard Number is locked. And multi numbers can be added by “,”, such as 911,999 Enable Password Dial When a number is entered beginning with the password prefix, the following N numbers after the password prefix will be displayed as *. N is the value entered in the Password Length field. For example: If the password prefix is 3 and the Password Length is 2, then dialing the number 34567 will display 3**67 on the phone. Password Dial Prefix Prefix for password dialing as described above. Password Dial Length Length for password dialing as described above. Enable Call History Allow phone to save missed call/dialed call/incoming call or not. Enable Default Line If enabled, you can assigned default SIP line for dialing out, not SIP1. Alllow IP Call Allow IP direct call, or disable IP call for dialing. Play Talking DTMF Allow DTMF voice played during talking, or only send DTMF without Tone local play. Ban Outgoing If enabled, no outgoing calls can be made. Enable Call Waiting If enabled, notifies user of a second call during a call. Caller ID of the new caller will be displayed. Press HOLD button to place existing call on hold and answer new call. ...
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outside call. If an intercom call is established, a second intercom call will be rejected. DND Return Code Specify SIP Code returned for DND. Default is 480 ‐ Temporarily Not Available. Busy Return Code Specify SIP Code returned for Busy. Default is 486 – Busy Here. Reject Return Code Specify SIP Code returned for Rejected call. Default is 603 – Decline. Active URI Limit IP IP address of the server for the Action URL messages described below. Push XML Server IP address for XML server which can send display content to the phone. Enable Call Waiting Enables audible notification of call waiting. Tone Enable Multi Line Enable phone to make calls for 10 lines max, or disable for 2 lines max. Enable Auto Switch Enable phone to select an available SIP line as default Line automatically. Play Dialing DTMF Tone Enable dialing DTMF tone played, or disable for dialing DTMF tone played. Action URL Settings URL for various actions performed by the phone. These actions are recorded and sent as xml files to the server. Sample format is ...
8.3.4.3 DIAL PLAN This phone supports 7 dialing modes: 17. Press "#" to Send– Dial the desired number, and press # to send it to the server. 18. Fixed Length – The number will be sent to the server after the specified number of digits are dialed. 19. Time Out – Number will be sent to the server after the specified time. 20. User Defined – Customized rules created by the user. There is a special feature in the dial plan for the case where the user must dial an access code to get an external line. A digit followed by a “,” will cause secondary dial tone to be generated. For example, assume a rule “9,xxxxxxx” is added. When the user dials 9, the phone will generate secondary dial tone. Then, when 8 digits have been dialed, they will all be sent to the server. 55/81...
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21. Press # to Do Blind Transfer ‐ Press # after entering the target number for the transfer. The phone will transfer the current call to the third party. 22. Blind Transfer on Onhook ‐ Hang up after entering the target number for the transfer. The phone will transfer the current call to the third party. 23. Attended Transfer on Onhook ‐ Hang up after the third party answers. The phone will transfer the current call to the third party. 24. Press DSS key to Do Blind Transfer – after talking, press DSS key, programed to memory key type, to make blind transfer directly. Dial Plan Special Characters Specifies a range of digits to match. May be a range, a list of ranges separated by commas, or a list of digits. * Match any single digit that is dialed. . Match any arbitrary number of digits including none. Tn A time out period before digits are sent of n seconds in length. n is mandatory and can have a value of 0 to 9 seconds. Tn must be the last 2 characters of a dial plan. If Tn is not specified it is assumed to be T0 by default on all dial plans. Cause extensions 1000‐8999 to be dialed immediately Cause 8 digit numbers beginning with 9 to be dialed immediately Cause 911 to be dialed immediately Cause 99 to be dialed after 4 seconds. Cause any number beginning with 9911 to be dialed 4 seconds after dialing ceases. 56/81...
Note: End with “#”, Fixed Length, Time out and Digital Map Table can be used simultaneously. 8.3.4.4 CONTACT Enter the name, phone number and ring type for each contact here. Field Name Explanation Phonebook Tables Group Dropdown box to select group Name Contact name Office Number, Mobile Contact phone numbers Number, Other Number Ring Type Ring type for this contact 57/81...
Group Contact group for this contact Add Contact Name Contact name Office Number, Mobile Contact phone numbers Number, Other Number Line Select line for associated contact number Ring Type Ring type for this contact Group Setting Choose the group or groups for this contact and move them to the Selected list on the right. Import Contact List Select File Click the browse button to select the phonebook file to import. Then click the update button and the selected file will be added to the phone. File must be xml, vcf or csv format. Export Contact List Export XML Export contacts to xml file. Export CSV Export contacts to csv file. Export VCF Export contacts to vcf file. Group Option Group Lists existing groups Name Enter name for new group Ring Type Ring type for group Blacklist Settings Type Select the blacklist type ‐ number or prefix Value Input number or prefix ...
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TFTP example for remote xml mode: Set the Phonebook Name as Linkcom ‐ Server URL is tftp://192.168.1.3/admin/phonebook/index.xml. Remote Phonebook Settings Phonebook Name Phonebook name displayed on the phone. Server URL Server url of the remote phonebook. SIP Line SIP line for the remote phonebook. User/password Authentication username and password. LDAP Settings Display Title The name displayed on the LCD. Version LDAP protocol version; supports 3 as default. Server Address LDAP server address. Server Port LDAP server port. Authentication depending on the server’s authentication mode. There are NONE,DIGEST‐MD5,CRAM‐MD5,SIMPLE Line Assigned SIP line for the LDAP dialing Username User name for LDAP authentication Password Password for LDAP authentication Search Base Search data directory Enable Calling Search Allow LDAP contact search and disply LDAP contact name during dialing and incoming call. Telephone Display LDAP contact’s telephone number 59/81...
Mobile Display LDAP contact’s mobile phone number Other Display LDAP contact’s ohter number Display Name Allow display LDAP contact name or not 8.3.4.6 WEB DIAL This feature allows a call to be initiated by a computer. To place a call, enter the number in the Dial Number box, select the line in the Line Selection box and press the Dial button. To end the call, press the Hangup button. 8.3.4.7 Multicast This feature allows you to make some kind of broadcast call to people who are in multicast group. You can configure a multicast key on the phone, which allows you to send a Real Time Transport Protocol (RTP) stream to the pre‐configured multicast address(es) without involving SIP signaling. You can also configure the phone to receive an RTP stream from pre‐configured multicast listening address(es) without involving SIP signaling. You can specify up to 10 multicast listening addresses. 60/81...
MCAST Settings Prority Define the priority of the active call, 1 is the highest priority, 10 is the lowest. Enable Page Priority The voice call in progress shall take precedence over all incoming paging calls. Name Listened multicast server name Host:port Listened multicast server’s multicast IP address and port. 8.3.5 Function Key 8.3.5.1 Softkeys Configure the functions performed by the softkeys under the LCD in various phone operating modes. 8.3.6 Maintenance 8.3.6.1 Auto Provision The phone supports PnP, DHCP, and Phone Flash to obtain configuration parameters. They will be queried in the following order when the phone boots. DHCP PnP server Phone Flash 61/81...
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Auto Provision Settings Field Name Explanation Current Config Version Show the current config file’s version. If the version of configuration downloaded is higher than this, the configuration will be upgraded. If the endpoints confirm the configuration by the Digest method, the configuration will not be upgraded unless it differs from the current configuration. Common Config Show the common config file’s version. If the configuration Version downloaded and this configuration are the same, the auto provision will stop. If the endpoints confirm the configuration by the Digest method, the configuration will not be upgraded unless it differs from the current configuration. CPE Serial Number Serial number of the phone User Username for configuration server. Used for FTP/HTTP/HTTPS. If this is blank the phone will use anonymous. Password Password for configuration server. Used for FTP/HTTP/HTTPS. Config Encryption Key Encryption key for the configuration file Common Config Encryption key for common configuration file Encryption Key Save Auto provision Save the Autoprovision username and password in the phone until Information the server url changes ...
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Plug and Play(pnp) Settings Enable PnP If this is enabled, the phone will send SIP SUBSCRIBE messages to a multicast address when it boots up. Any SIP server understanding that message will reply with a SIP NOTIFY message containing the Auto Provisioning Server URL where the phones can request their configuration. PnP Server PnP Server Address PnP Port PnP Server Port PnP Transport PnP Transfer protocol – UDP or TCP PnP Interval Interval time for querying PnP server. Default is 1 hour. Phone Flash Settings Server Address Set FTP/TFTP/HTTP server IP address for auto update. The address can be an IP address or Domain name with subdirectory. Protocol Type Specify the Protocol type FTP, TFTP or HTTP. Config File Name Specify configuration file name. The phone will use its MAC ID as the config file name if this is blank. Update Interval Specify the update interval time. Default is 1 hour. Update Mode 1. Disable – no update 2. Update after reboot – update only after reboot. 3. Update at time interval – update at periodic update interval 64/81...
TR069 Settings Enable TR069 Enable/Disable TR069 configuration ACS Server Type Select Common or CTC ACS Server Type. ACS Server URL ACS Server URL. ACS User User name for ACS. ACS Password ACS Password. TR069 Auto Login Enable/Disable TR069 Auto Login. "Inform" Sending Period Time between transmissions of “Inform” Unit is seconds. 8.3.6.2 Syslog Syslog is a protocol used to record log messages using a client/server mechanism. The Syslog server receives the messages from clients, and classifies them based on priority and type. Then these messages will be written into a log by rules which the administrator has configured. There are 8 levels of debug information. Level Name Description 0 Emergency System is unusable. This is the highest debug info level. 1 Alert Action must be taken immediately. 2 Critical ...
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Syslog Configuration Field Name Explanation Syslog Settings Server IP Syslog server IP address. Server Port Syslog server port. MGR Log Level Set the level of MGR log. SIP Log Level Set the level of SIP log. IAX2 Log Level Set the level of IAX2 log. Enable Syslog Enable or disable syslog. Watch Dog Enable watch Dog Enable watchdog phone die chance to automatically restart, disable watchdog does not Web Capture Start Capture a packet stream from the phone. This is normally used to troubleshoot problems. Stop Stop capturing the packet stream 66/81...
8.3.6.3 Config Setting Config Setting Field Name Explanation Save Configuration Save the current phone configuration. Clicking this saves all configuration changes and makes them effective immediately. Backup Configuration Save the phone configuration to a txt or xml file. Please note to Right click on the choice and then choose “Save Link As.” Clear Configuration Logged in as Admin, this will restore factory default and remove all configuration information. Logged in as Guest, this will reset all configuration information except for VoIP accounts (SIP1‐2 and IAX2) and version number. 67/81...
8.3.6.4 Update This page allows uploading configuration files to the phone. Update Field Name Explanation Web Update Browse to the config file, and press Update to load it to the phone. Various types of files can be loaded here including firmware, ring tones, local phonebook and config files in either text or xml format. Web Update TFTP/FTP Update FTP/TFTP server address for download/upload. The address can be Server Address IP address or Domain name with subdirectory. FTP server Username for download/upload. User FTP server password for download/upload. Password 68/81...
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Name of update file or config file. The default name is the MAC of File name the phone. Note: The exported config file can be modified. The config file is made up of modules. Modules which do not need changes may be deleted. For example, a config file can be downloaded and all modules removed except the SIP module. After rebooting, only the SIP settings will be changed. Type Action to be executed by the phone. 1. Application update ‐ download system update file 2. Config file export ‐ Upload config file to FTP/TFTP server. It can then be named and saved. 3. Config file import ‐ Download the config file from FTP/TFTP server. The configuration will be effective after the phone is reset. 4. Phone book export (.vcf, .csv, .xml) ‐ Upload the phonebook file to FTP/TFTP server. It can then be named and saved. 5. PhoneBook import (.vcf, .csv, .xml) ‐ Download phonebook file from FTP/TFTP server. Protocol Select FTP/TFTP server. Update Logo File Select File ...
8.3.6.5 Access User accounts can be added or deleted from this page. The authority of accounts can also be changed. Access Configuration Field Name Explanation LCD Menu Password Settings Menu Password Sets the password for entering the setup menu from the phone keypad. The password must be only digits. Keyboard Lock Settings Quick lock key code Direct press set passwords can lock the keyboard The key password Set phone keyboard lock password, must enter Numbers, limit is no more than 6 characters Open the keyboard lock Set whether to open the keyboard lock, default to cancel User Settings This table shows the current user accounts Add User User Set User Account name 70/81...
User Level There are two levels. Root user can modify the configuration. General user can only read the configuration. Password Set the password Confirm Confirm the password User Management Select the account and click Modify to modify the selected account. Click Delete to delete the selected account. A General user can only add another General user. 8.3.6.6 Reboot Some configuration modifications require a reboot to become effective. Clicking the Reboot button will cause the phone to reboot immediately. Note: Be sure to save the configuration before rebooting. 8.3.7 Security 8.3.7.1 WEB FILTER 71/81...
WEB Filter The Web filter is used to limit access to the phone. When the web filter is enabled, only the IP addresses between the start IP and end IP can access the phone. Field Name Explanation Start IP Address Beginning IP Address for MMI Filter End IP Address Ending IP Address for MMI Filter Add Add this filter range to the Web Filter Table Enable Web Filter Select to enable MMI Filter. Apply Make filter settings effective. Note: Once a range is added, it can be modified or deleted. Note: Be sure that the filter range includes the IP address of the configuration computer. 8.3.7.2 Firewall Firewall Configuration Firewall rules can be used to prevent unauthorized Internet users from accessing private networks connected to this phone (input rule), or prevent unauthorized devices connected to this phone from accessing the Internet (output rule). Each rule type supports a maximum of 10 items. Field Name Explanation Enable Input Rules Enable rules limiting access from the Internet. 72/81...
Enable Output Rules Enable rules limiting access to the Internet. Input/Output Specify if the current rule is input or output. Deny/Permit Specify if the current rule is Deny or Permit. Protocol Filter protocol type (TCP/ UDP/ ICMP/ IP) Port Range Set the filter Port range Src Address Set source address. It can be a single IP address or use * as a wild card. For example: 192.168.1.14 or *.*.*.14. Dest Address Set destination address. It can be a single IP address or use * as a wild card. For example: 192.168.1.14 or *.*.*.14. Src Mask source address mask. For example: 255.255.255.255 points Set the to one host while 255.255.255.0 points to a C type network. Dest Mask destination address mask. For example: 255.255.255.255 Set the points to one host while 255.255.255.0 points to a C type network. When a connected device tries to access 192.168.1.118, the phone will deny the request because of the out_access rule. Access to any other IP address will be allowed. Click the Delete button to delete the selected rule. 8.3.7.3 Network Address Translation (NAT) NAT is the process of modifying IP address and port information in transition from a private to a public network. NAT allows the use of one public address to support many private ...
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DMZ Configuration Servers in a network most vulnerable to attack are those which provide services to users outside the local network. Many times these computers are placed into their own sub‐network to provide more protection to the rest of the local network. This sub‐network is called a DMZ (taken from “demilitarized zone”). Computers in the DMZ have limited connectivity to specific hosts in the internal network, although communication with other hosts in the DMZ and to the external network is allowed. This allows hosts in the DMZ to provide services to both the internal and external network, while a firewall controls the traffic between the DMZ servers and the internal network clients. . The following chart describes the network access control of DMZ 74/81...
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Application Layer Gateway (ALG) Settings Field Name Explanation IPSec ALG Enable/Disable IPSec encryption. Default is enabled. FTP ALG Allow the ALG to securely pass FTP traffic. Default is enabled. PPTP ALG Allow the ALG to securely pass PPTP traffic. Default is enabled. Network Address Translation (NAT) Table Shows the NAT TCP and UDP mapping tables 75/81...
NAT Table Option Transfer Type Select the TCP or UDP protocol. Inside IP Set the local IP address of device. Inside Port Set the LAN (inside) port for NAT mapping Outside Port Set the WAN (outside) port for NAT mapping Note: After entering settings, click the Add button to add new mapping table data. To delete an entry, enter its information and then click the Delete button. Notice: The phone supports 10M/100M adaptive. Under loaded conditions traffic through the phone NAT may not reach 100M. 8.3.7.4 VPN The phone supports remote connection via VPN. It supports both Layer 2 Tunneling Protocol (L2TP) and OpenVPN protocol. This allows users at remote locations on the public network to make secure connections to local networks. Field Name Explanation VPN Status Shows the current VPN IP address. VPN Mode Select L2TP. You can choose only one for current state. After you select it, save the configuration and reboot the phone. Enable VPN Enable/Disable VPN. L2TP Select Layer 2 Tunneling Protocol OpenVPN ...
VPN User Set User Name access to VPN L2TP Server. VPN Password Set Password access to VPN L2TP Server. 8.3.7.5 Security Field Name Explanation Update Security File Select Security File Browse to the security file to be updated. Click the Update button to update. Delete Security File Select Security File Select the security file to be deleted. Click the Delete button to Delete. SIP TLS File Show SIP TLS authentication certificate. HTTPS File Show HTTPS authentication certificate. OpenVPN Files Show OpenVPN File authentication certificate file. 8.3.8 Logout Click Logout to exit the phone web page. 77/81...
Appendix Specification 9.1.1 Hardware Item Specification Power Adapter Input: 100‐240V Output: 5V 1A WAN 10/100Base‐ T RJ‐45 1 PORT Port LAN 10/100Base‐ T RJ‐45 1 PORT Power Consumption Idle: 2.5W Active: 2.8W LCD Size 128x48 pixels Operation Temperature 0~40 Relative Humidity 10~65% CPU Broadcom SDRAM 16MB Flash 4MB Dimension(L x W x H) 250×205×60mm Weight 0.84kg 9.1.2 Voice Features ...
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SIP support SIP domain SIP authentication none basic MD5 DNS Peer to Peer/ IP call Automatic line selection 9 Standard ring tones and 3 user‐defined ring tones DTMF SIP info DTMF In‐Band RFC2833 AUTO SIP applications Call Forward Call Transfer(Blind/Attended) ...
Support call logs Incoming Calls Outgoing Calls Missed Calls Max of 300 Records Each Supports vCard/XML/CSV Support IAX2 Programmable Soft Keys Code synchronization IP PBX IMS Supports Click to Dial via Web Phone Book Keypad Lock with Emergency Call Customized LCD logo as screensaver Ring Tone via Speaker Customized Signal Tone Parameters ...
Digit‐character map table Keypad Character Keypad Character 7 P Q R S p q r s 1 @ 2 A B C a b c 8 T U V t u v 3 D E F d e f 9 W X Y Z w x y z 4 G H I g h i */. 5 J K L j k l 0 6 M N O m n o #/= 82/81...